So, I was thinking.. During 80ties when processors and rack devices started to emerge, lots of players embraced them - and you can definitely hear that. Sound became thinner, everything sounding more or less processed.

How bout today? Do you know any awesome guitar player that sounds "natural", that is using digital gear?

What is a digital processor ?Just copying analogy of course . But i have read somewhere that digital signal processors has brought two things only to the guitar players: pitch-shifting (harmonizer) and whammy pedal ! So true In that case, I know them a lot !

Live sound is perhaps a little different to anything that is recorded, mixed and mastered for commercial release. Any commerical release is almost certain to have been through an analogue chain at some stage and that willl affect the sound quite significantly. Very few professional sound engineers rely on digital processing. At mastering, for instance, the majority of processing is done on hardware rather than on digital plug ins/emulations. Simple reason is that so far very few, if any, come close to acheiving the same sound as good hardware for lots of reasons. This includes vsts that emulate or model hardware. Digital effects/processing may be used in a studio by a guitar player but the extent to which that that is then affected/enhanced by the use of analogue outboard arguably is quite significant.

What works in digital imho are time and harmonising type effects. What is much better done in hardware tends to be dynamics, eq and distortion/saturation.

I know bands that have switched to using the Fractal Axe FX Ultra for gigs, as in a live setting, it's hard to tell the difference. Also, all the effects being in one place saves them carrying around all their pedals and other gear. It's just way more convenient.

Digital is digital, but my Digitech RP 1000 gives me access to many different amps, effects, stompboxes, cabs etc that I otherwise would never have access to because of expense. Lots of bedroom musicians have made some great recordings on youtube thanks to some of these devices. Many guitarists use them on stage live at their gigs. The guitars on this video (link) were recorded using a Boss GT 10 floor processor.

Yes unfortunately I agree with all my heart about digital not really cutting it (although I havent tried the Axe fx).

It actually feels like my line6 M13 ruins my chain as the only digital component... heck 13 is for bad luck! No seriously I like the sounds of it - but it seems that it affects the chain negatively somehow. It's probably the A/D that does it..

With converters, specially low quality ones that they usually put in ALL processors, the difference is huge even on bypass. It removes all the nice top end sizzle, and it disables the possibility for a good quality feedback. Digital is more practical tho..

I've seen Dethklok and Trans-Siberian Orchestra and both of them have a guitarist that use a Pod XT.

Dethklok ran it into a Krank amp when I seen them live so it still had some warmth to it but when they put it into leads and a hi-gain distortion it got less warm and went more towards a sterile sound.

TSO I believe is ran straight into a mixer, honestly I can't tell, the quality of the show itself is about the same as seeing some one lip sync a cd, its spot on.

I have had digital all my life, old school DOD, Zoom 505-II, Pod XT, Boss GT-10, Pod HD500. They have always done what I needed for my budget. Look at all the instructors that use guitar rig for lessons, they sound amazing. I think that there is always a place for it somewhere. You can always add cab sims and such that help make it more realistic and gives more depth to the sound.

You can sit on youtube listening to fearedse test amps, stomps, and multis. Digital sound Axe FX to my ears is the closest sound to the real thing. With having the Pod HD500 myself and using the rectifier tone it got the job done. But truthfully it has nothing on the amp.

Unfortunately for me I have never heard a guitarist in person not use something digital. So I do not have a benchmark to compare against. That said, I always feel my tone is not where it could be and I wonder if that is just something internal in me that wants to hear more than what my processor gives... ?????

Periphery use Axe FX for all their recordings, and it sounds stunning. Cynic also use Axe Fx and I think they have easily one of the best clean sounds to ever come out of a guitar.

Right now, analog is better in someway where digital is not, and vice versa for digital. Soon though, I believe digital will be indistinguishable from analog (two decades or something?). As for the mastering part of it like Tony mentioned, hard to say when digital will catch up in that respect.

Seeing how the main means of listening to music now is the MP3, do you guys think it matters if the guitars are recorded with a real amp or a simulator? I'm no expert on recorded audio, but I do notice that MP3's have a bit more brittleness than CD's. I'm under the assumption its because MP3's are compressed files.

But there is one thing that I found that is almost always missing from digital amps and effects: the X-factor. It just can't recreate electrons flowing through a circuit. A live current is exactly that, something that is alive. An analog system is the harnessing and taming of electricity, its like a lion on a leash. Ones and zeros can't compensate for the infinite variability in waves and currents that is provided by the nature of an analog system... at least not yet

But I do like the clean tones provided by digital systems. I like them because they are SUPER clean. Analog cleans, especially from a Fender amp, are fantastic as well. I don't think one is better than the other. Just each one has different applications.

This post has been edited by Mudbone: Aug 15 2011, 12:35 AM

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He who laughs last thinks slowest.

"That which can be asserted without evidence, can be dismissed without evidence." - Christopher Hitchens

......What works in digital imho are time and harmonising type effects. What is much better done in hardware tends to be dynamics, eq and distortion/saturation.

Tony is spot on per usual As in most cases, money plays a bit part in how much Analogue/Boutique gear gets usedduring production and post. That stuff ain't cheap. You find both in use at all ends of the spectrum of course, but highdollar shops, studios, bands, do tend to have some actual gear involved in the signal chain to "warm up" the signal. For examplea guitarist might do his tracks at home through this AXE FX, even for a big band, but his track might get put through and old analogue console/preamps/compressors, during mix down. Then more live circuits during Mastering.

The big gap between Digital and Analogue, is (IMHO, and mostly) about sampling (believe it or not) the stuff we can't hearimpacts the stuff we can hear. Things like Harmonic Distortion, even at inaudble frequencies, create resonance/harmonics/noise/warmth, in the frequencies our ears actually pick up. So (HERE IS WHERE IT GETS TECHNICAL) if we are using the Nyquist theorem, and our max frequency is half our sampling rate, and we work at 48k, then 24k is our highesttone. We can't hear much beyond 18 or 20k, but again it's not the frequencies we can hear that are the point here.It's about the stuff that gets chopped off during sampling/quantization.

So the new trend in digital gear is (wait for it) Hyper Oversampling! This way we start to include the bits we can't hear in our processing. So technology is on the right track. But until we can sample in crazy numbers like, oh, 968k+ (Which is science fiction at this point) we can't sample adequate information to trick the trained ear.

GIven the collapse of CPU/Ram/storage price curves over the last decade (if cars kept pace we'd have cars that could fly and cost pennies) and the development of software on a tear, I think we are going to get close than we have ever been very very soon, in terms of making a sampled, digital signal have the warmth and love/heat/magic that pure analogue signal chain has now.

Sampling rate of a 96KHz is already high enough, but the problem lies in the usage of low pass antialiasing filters. On high end converters, filters are sharp and precise enough even on high sampling rates, and this is what gives quality to those units.

The big gap between Digital and Analogue, is (IMHO, and mostly) about sampling (believe it or not) the stuff we can't hearimpacts the stuff we can hear. Things like Harmonic Distortion, even at inaudble frequencies, create resonance/harmonics/noise/warmth, in the frequencies our ears actually pick up. So (HERE IS WHERE IT GETS TECHNICAL) if we are using the Nyquist theorem, and our max frequency is half our sampling rate, and we work at 48k, then 24k is our highesttone. We can't hear much beyond 18 or 20k, but again it's not the frequencies we can hear that are the point here.It's about the stuff that gets chopped off during sampling/quantization.

So the new trend in digital gear is (wait for it) Hyper Oversampling! This way we start to include the bits we can't hear in our processing. So technology is on the right track. But until we can sample in crazy numbers like, oh, 968k+ (Which is science fiction at this point) we can't sample adequate information to trick the trained ear.

GIven the collapse of CPU/Ram/storage price curves over the last decade (if cars kept pace we'd have cars that could fly and cost pennies) and the development of software on a tear, I think we are going to get close than we have ever been very very soon, in terms of making a sampled, digital signal have the warmth and love/heat/magic that pure analogue signal chain has now.

Yes to some extent. The Nyquist point at 24kHz when the sampling is at 48 is above our audible range but it can still be an issue due to aliasing and folding back of that down in to the upper end of our audible range. To some extent though many of the non high end DACs aren't good enough though to seriously reproduce much above 16-18kHz with any sense of accuracy and without the low level dropping down to the noise floor or below. So the folding back, whilst there, may generally go unnoticed.

At the present moment in time Ivan's right:

QUOTE (Ivan Milenkovic @ Aug 15 2011, 02:27 PM)

Sampling rate of a 96KHz is already high enough, but the problem lies in the usage of low pass antialiasing filters. On high end converters, filters are sharp and precise enough even on high sampling rates, and this is what gives quality to those units.

lbsolutely Ivan abeit that there is another reason why 96kHz is more than sufficient. Quite simply above 96kHz you generate a huge amount of data and that can cause problems as the hardware struggles to process it.

At the moment you'd be hard pushed to find any professional mastering studio that routinely runs at anything other than 44.1/48kHz and 24 bit. That's even though we use high end convertors at least in part for the reasons Ivan describes. By high end convertors I really mean ones from likes of Lavry, Cranesong, Forsell, Prism etc than the prosumer stuff and below.

Mudbone - yes mp3s are (over) compressed and the compression is part of the problem. That compression is made even worse if apoorly encoded mp3 is then processed further for, for example, radio or Youtube broadcast. You then run in to an affect called cascading where the end result is a very thin and brittle audio.

Another (related) issue is that the encoding to mp3 also introduces other issues related to innaccurate internal summing when the mp3 is transcoded that can drive transients in to clipping. These intersample overs can't be properly interpreted by the DAC and ends up being reproduced as high frequency distortion and or drop out.

As a mastering engineer we have to pay attention to issues like this and account for whether the final product is likely to be digital distribution as mp3s or hard copy CDs and respond accordingly. That's part of the quality assurance that we provide. What sadly is more often the case nowadays is that a pro ME isn't used. The person who does the work either is unaware of these issues, doesn't understand the science, doesn't have the correct equipment or doesn't care enough to deal with it. You then end up with the nasty, brittle mp3s that you hear so much everywhere. Sad really,musicians spend years learning to play, take time to write their music and practice it. They go in to a studio and record it but for the sake of a few dollars they don't have it mastered professionally and end up with...

Tony is correct. Theoretically one can digitize a signal and lose nothing, but putting theory into practice is another thing.

This goes back to when I studied computer science, this is the math that it primarily is built on. I had to compete with Chinese and Indian students in this class, meaning I only got though by visiting the professor during office hours on a regular basis, and prostrating myself on the ground and worshiping him, and begging for mercy.

If you want to see the gory details of what type of course you NEVER want to take competing for grades with nearly all Indian and Chinese students, here you go.

Yes to some extent. The Nyquist point at 24kHz when the sampling is at 48 is above our audible range but it can still be an issue due to aliasing and folding back of that down in to the upper end of our audible range. To some extent though many of the non high end DACs aren't good enough though to seriously reproduce much above 16-18kHz with any sense of accuracy and without the low level dropping down to the noise floor or below. So the folding back, whilst there, may generally go unnoticed.

At the present moment in time Ivan's right:

lbsolutely Ivan abeit that there is another reason why 96kHz is more than sufficient. Quite simply above 96kHz you generate a huge amount of data and that can cause problems as the hardware struggles to process it.

At the moment you'd be hard pushed to find any professional mastering studio that routinely runs at anything other than 44.1/48kHz and 24 bit. That's even though we use high end convertors at least in part for the reasons Ivan describes. By high end convertors I really mean ones from likes of Lavry, Cranesong, Forsell, Prism etc than the prosumer stuff and below.

Mudbone - yes mp3s are (over) compressed and the compression is part of the problem. That compression is made even worse if apoorly encoded mp3 is then processed further for, for example, radio or Youtube broadcast. You then run in to an affect called cascading where the end result is a very thin and brittle audio.

Another (related) issue is that the encoding to mp3 also introduces other issues related to innaccurate internal summing when the mp3 is transcoded that can drive transients in to clipping. These intersample overs can't be properly interpreted by the DAC and ends up being reproduced as high frequency distortion and or drop out.

As a mastering engineer we have to pay attention to issues like this and account for whether the final product is likely to be digital distribution as mp3s or hard copy CDs and respond accordingly. That's part of the quality assurance that we provide. What sadly is more often the case nowadays is that a pro ME isn't used. The person who does the work either is unaware of these issues, doesn't understand the science, doesn't have the correct equipment or doesn't care enough to deal with it. You then end up with the nasty, brittle mp3s that you hear so much everywhere. Sad really,musicians spend years learning to play, take time to write their music and practice it. They go in to a studio and record it but for the sake of a few dollars they don't have it mastered professionally and end up with...

Sampling rate of a 96KHz is already high enough, but the problem lies in the usage of low pass antialiasing filters. On high end converters, filters are sharp and precise enough even on high sampling rates, and this is what gives quality to those units.

96k is "enough" for decent digital recordings sure I agree with everyone on this. And with Ivan on the issues relating to converters, and with Tony, that capturing more requires MASSIVE data. But what I"m more talking about here is "warmth/analogue magic" that gets clipped off even at 96k. At 96k your highest frequency is some where around 45k, which is much better than 48k, but still nowhere close to being able to capture the full range of harmonic distortion happening in a true analogue signal. As I was talking about in the last post, the stuff we can't hear, impacts the stuff we can hear so its about capturing things are are not audible. Sub harmonics (Below 20hz and Hyper/super Harmonics above 20k) provide "color" and "depth" to actual signals that happen in the real world. Like a guitar through an amp. There are resonant frequencies that we will never hear coming out of a Marshall stack that interact with the audible range to provide the "sound".

We will need to see crazy high sample rates and over sampling, with better aliasing, bit depth etc. in order to get our digital systems to have the "Analogue Warmth" that so many of us try to put back in our signal path with plugins, 2 track tape, running through an analogue board etc. Then better DAC, etc.

FKALICH: (Hardware limitations) This just seems impossible now. In five years it won't be. Moore's Law of MicroProcessor advancement marches on. Hardware gets fasters, bus width gets bigger, hard drives get larger, etc. It's just a matter of time. We will wonder how we ever got by with these low sample rates. I remember when 44.1 was "Enough" for most folks and 48k was high end. Now it's 96. Give it a few years.

Here is a abstract to an article talking about the effect I'm on about here. It's called the "Hypersonic Effect". Essentially, encoding and reproducing sounds well beyond the audible limit of most hardware are pleasing to the ear much like a real instrument. It's still a new area of research in the Audio Engineering Society, even though the initial research goes back more than 10 years. This is from the